9.

SIP

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9.1 Introduction (1/2) ..........................................................................3

9.1 Introduction (2/2) ..........................................................................4
9.2 Addressing ...................................................................................5
9.3 Determing a SIP Server................................................................6
9.4 SIP Transactions ..........................................................................7
9.5 SIP Call Signaling.........................................................................8
9.6 Location Server ............................................................................9
9.7 SIP in Proxy Mode (1/2).............................................................. 10
9.7 SIP in Proxy Mode (2/2).............................................................. 11
9.8 SIP in Redirect Mode.................................................................. 12
9.9 SIP in Comparison to H.323 (1/5)............................................... 13
9.9 SIP in Comparison to H.323 (2/5)............................................... 14
9.9 SIP in Comparison to H.323 (3/5)............................................... 15
9.9 SIP in Comparison to H.323 (4/5)............................................... 16
9.9 SIP in Comparison to H.323 (5/5)............................................... 17

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9.1 Introduction (1/2)

When establishing an IP based telephone system, alongside the infrastructure two things
must be taken into consideration. First, there must be a technique for the transmission of
multimedia data over an IP network. This will usually be voice data, but video transmission
will play a growing role. Second, a set of protocols is needed to implement different signaling
and control mechanisms.
Two different methods have become established in IP telephony. The ITU specified H.323,
an attempt to adapt the principles of traditional circuit switched network handling. Signaling
for call set-up and tear-down is based on the Q.931 standard also used in ISDN.
The IETF set out to create an alternative signaling system for IP telephony with the Session
Initiation Protocol (SIP). Their motivation was to use the opportunities offered by the internet,
instead of adapting old concepts. Specifically, the characteristics of HTTP and SMTP
provided the base for SIP.
The only common factor shared by H.323 and SIP is that they both use RTP for real time
data transmission.

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9.1 Introduction (2/2)

SIP is a signaling protocol located which resides on the application level. It is able to setup
and tear down multimedia sessions, ranging from simple phone calls to conference calls with
multiple users. SIP is completely independent from the underlying transport protocol, thus
making it suitable for use outside IP networks. Alongside session establishment, SIP
supports different signaling tasks such as inviting additional users to an existing session. It
can also be used for rerouting signals, a function that is used to recreate certain ISDN
services.
Another important concept in SIP is Personal Mobility.

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9.2 Addressing

All objects which can be addressed by SIP are identified using an SIP Uniform Resource
Locator (URL). The format sip:user@host - which is basically comparable to an e-mail
address. The user part of the URL address can be chosen freely - it can be a personal name,
a phone number, a group of persons or can refer to a common service which is not assigned
to a specific person. It is not particularly important which SIP-URL a user chooses, but
remember that a name-based URL (for example sip:max.meyer@mustermann.com) does
not provide the anonymity of a phone number. In contrast to standard phone networks,
undesired and unauthorized calls can be blocked. SIP offers various mechanisms for
authentication and access control.

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9.3 Determing a SIP Server

As the transmission of requests in SIP follows HTTP very closely, the same definitions of
client and server are used. A terminal which sends a request to another is thus the client, the
receiver is the server. SIP is not an extension or enhancement to HTTP but an independent
protocol which happens to have many similarities to HTTP. There are two ways to send a
request from a client to a server. In the first, simpler option - called Proxy Mode - the request
is sent to a local SIP proxy which takes care of further routing. In the second option - called
Redirect Mode - the request is sent directly to the server, which has been determined in
advance by analyzing the request.

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9.4 SIP Transactions

After establishing the connection to the server, the client may send one or more requests.
The server answers with one or more responses. Both requests and responses are
described as SIP transactions. The client may decide which protocol - TCP or UDP - to use
in handling a transaction. If TCP is used for a request, it will also be used for the response. A
separate TCP connection is needed for each transaction. Optionally, a number of requests
can be transmitted within one TCP connection. Alternatively UDP can be used, making it
possible to address multiple servers in a multicast operation - a function which is not possible
with TCP. Unfortunately UDP does not offer a secure data transport. Thus both server and
client have to support mechanisms for the resending of unacknowledged requests.

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9.5 SIP Call Signaling

To enter a session, an SIP invitation consisting of two requests is needed - an "invite"

followed by an "acknowledge". The "invite request" is a request to join the session. It
contains all the information needed to join the session, described using the Session
Description Protocol (SDP). After the called party has acknowledged the invite request with
"OK 200", it opens a RTP media channel for real time data transmission using UDP. The
calling party also opens a RTP media channel and ends the transaction by sending an "ACK
request". This second request confirms that the acknowledgement "OK 200" has been
received. To leave a session, the subscriber sends a "Bye request". The entire scenario is
very similiar to the 3-way handshaking used in TCP connection setup.

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9.6 Location Server

Personal mobility can mean that the subscriber's location may change. SIP does not use a
central server to store the locations of subscribers. Indeed, SIP is not able to locate a
particular subscriber. Instead, it relies on a location service for this task.
Two methods are available. In the first method, the location service searches for the
subscriber independently, using protocols like the Lightweight Directory Access Protocol
(LDAP).
The second method is a subscriber's discretion. Every subscriber is able to register his
current location in a SIP proxy by using the "register request" command.

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9.7 SIP in Proxy Mode (1/2)

The call can be handeled in two different ways: proxy mode and redirect mode. In Proxy
Mode, the proxy server takes care of call control, handling the terminal requests. The
addresses come from the Location Service. The example shows a session establishment in
eight steps. The first step is an "invite request" containing the address of the subscriber who
is being invited to the session. In this case it is mayer@mustermann.com. DNS is used to
find the destination address: in this example it is a SIP proxy of the "mustermann.com"
domain. As this server has no information on the location of the terminal being invited, the
second step is to contact the location service. This service attempts to locate the subscriber
'mayer'. If this is successful, the third step is the transfer of the information needed to forward
the request - here, it would be 'mayer@ 195.100.33.33'.
In step four, the "invite request" is sent to the target PC where the invited subscriber is
located

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9.7 SIP in Proxy Mode (2/2)

If the subscriber accepts the "invite request" an acknowledgement "ok 200" is sent in step
five, and forwarded to the sender of the "invite request" in step six.
As this acknowledgement has to be re-acknowledged, an "ack request" is sent to the SIP
proxy in step seven.
The proxy still has the location information, and so the "ack request" is forwarded to the
invited subscriber directly (without contacting the location service) in step eight. In this step,
the media channels are also established. If a subscriber wants to leave the session, a "bye
request" is sent directly.

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9.8 SIP in Redirect Mode

In Redirect Mode the SIP server is asked to localize the subscriber in question. This request
is not passed on to the subscriber being sought, instead the SIP server returns the
addresses of the subscribers the requesting party must contact the other subscriber directly.
The Redirect Server accepts the "invite request" to 'mayer@mustermann.com' in step one
and contacts the location server in step two. After the location server has found user 'mayer',
the current address 'mayer@home.de' is returned to the redirect server in step three.
In step four, the address information is sent from the Redirect Server to the calling party. In
step five the user agent sends an "ack request" to the redirect server, acknowledging the
transaction. In step six, the calling party's user agent sends an "invite request" directly to the
address given by the redirect server. If the call is accepted, the called subscriber sends back
an "ok 200" acknowledgement as step seven. This acknowledgement must be reacknowledged, so in step eight the session initiator sends an "ack request" to the called
party.

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9.9 SIP in Comparison to H.323 (1/5)

H.323 and SIP represent two completely different approaches to IP telephony although both
use RTP for the transmission of multimedia data.
IP telephony requires a signaling mechanism. In H.323 version 2, this signaling mechanism
is based on a reliable transport protocol like TCP. Thus two phases are needed for call
setup. One is the setting up of the TCP connection; the other is the transmission of the
signaling message for setting up the call itself. It has only been possible to use UDP as the
signaling canal since version 3 of H.323. In connection setup signaling, H.323 is in many
ways equivalent to SIP. Call release is also quite similar as the "release complete" in H.323
is equivalent to the "bye request" in SIP.

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9.9 SIP in Comparison to H.323 (2/5)

Many other protocols are involved in the specification of H.323. They include H.225 for
signaling, H.245 for various control tasks, H.450 for additional service specification, H.235 for
security and ciphering aspects, and H.246 for service interoperability with telephone
networks. Thus H.323 is quite complex - for example for call forwarding parts of H.450,
H.225, and H.245 are needed.
Another difference between the two systems is the number of elements defined. H.323
contains several hundred elements, while SIP only has 37 different headers.
The syntax of H.323 messages is expressed in binary; all SIP messages are in text format.
When troubleshooting, the text format is much easier to handle as it can be read without any
conversion.

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9.9 SIP in Comparison to H.323 (3/5)

H.323 was originally designed for use in LANs. Where scalability via name resolution with
DNS is concerned, even the newest versions of H.323 do not provide an efficient mechanism
for locating people in other domains - in contrast to SIP. Server load is one more aspect
which can be used to measure scalability. In SIP the signaling is state-independent and can
use UDP if necessary. This reduction of the protocol overhead is a source of high scalability,
especially for backbone servers. In contrast, an H.323 gatekeeper needs a dedicated TCP
session for each separate signaling channel.
In future versions of H.323 it will be possible to use UDP as an alternative signaling carrier.
As the gatekeeper will have to oversee the call state throughout the entire call, there will be
scaling problems in bigger systems.

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9.9 SIP in Comparison to H.323 (4/5)

Both H.323 and SIP offer a wide variety of services alongside normal speech calls. It is
difficult to compare the two systems as both of them regularly offer new services.
Nevertheless, one aspect of each system can be emphasized.
H.323 offers numerous services concerned with capability exchange. H.323-terminals can
describe in detail which codecs they can use with which parameters. Furthermore H.245 can
be used to perform a handshake, negotiating which codec to use.
SIP is restricted to sending a list of supported codecs to the communication partner, which
decides which codec to use. A complete handshake does not take place. On the other hand,
SIP offers more functions for personal mobility as it was conceived with these functions in
mind.

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9.9 SIP in Comparison to H.323 (5/5)

Telecommunication providers are interested in the commercial use of IP telephony. From this
point of view H.323 is the best solution. Using an H.323 gatekeeper, various services can be
offered, but their use is restricted to customers who have been successfully registered.
Access control can also be handled by the gatekeeper, making it possible to track the
subscribers with regards to the type, time and duration of the services used. This information
forms the base for charging.
In SIP the gatekeeper has no equivalent. In addition, in SIP signaling is decentralized, which
makes it very difficult to develop commercial applications. In its current specification, SIP
does not play an important role for providers.

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